Much research has been conducted in recent years in an effort to adapt digital signal processing techniques for use in communication systems. These research efforts have been spurred by the development of the digital signal processor (DSP), which, as is well-known in the art, is an integrated circuit similar to a microprocessor that allows mathematical operations crucial to signal processing applications to be performed at very high speeds. The use of a DSP vastly simplifies the realization of finite impulse response filters, for example, and computation of Fast Fourier Transforms can now be done at speeds unprecedented for small, integrated circuit devices.
The principal attraction of DSP's to designers is flexibility. Since a DSP is a software-driven device, it is a relatively simple matter to change the entire complexion of a DSP-based communication unit simply by changing software. For example, an FM (frequency modulation) system can be changed to an AM (amplitude modulation) system by re-programming DSP ROM (read only memory).
The use of a DSP also means increased flexibility in terms of communication system features. New features can be added quickly and easily by reprogramming, and the latest improvements in signal processing algorithms can be added to improve system performance.
Perhaps most importantly, designers are not limited by considerations of physical realizability that hamper the designers of analog systems. If a filter or other function can be expressed mathematically and executed in a DSP (within any constraints on execution time dictated by the nature of the system in question, of course), the designer simply programs the function into the DSP without having to worry about selection of the proper resistors, capacitors, inductors, or other components that would be part of a physical system. A DSP-implemented filter is also infinitely repeatable in its performance, unlike physical filters, which are affected by factors such as aging and temperature.
Since DSP-based systems are inherently very flexible, the task of determining which of many modulation and transmission protocols best fits a particular application is greatly simplified. Very complex systems may be implemented with relative ease, thus helping to ensure that system performance is maximized.
One of the problems faced by any RF (radio frequency) communication system designer is that of frequency-selective fading. This phenomenon results from reflection of transmitted signals and the interaction of these reflections as they arrive at the receiver. Each reflected signal has its own delay and phase, and, when the reflections interact with the original transmitted signal, inter-symbol interference is the usual result, at least in a data system. This type of interference makes proper interpretation of transmitted signals difficult.
The use of multiple sub-channels helps to counteract the effects of frequency-selective fading. Each sub-channel is formed by modulating a sub-carrier signal with some portion of the information signals to be transmitted. Then, a composite signal is formed by combining these sub-channels, and the composite signal is used to modulate a primary carrier.
In one such system, sixteen symbol quadrature amplitude modulation, or 16-QAM, is used to transmit information signals arranged into TDM time slots, with each time slot having sync and pilot symbols inserted in a predetermined pattern among the data symbols to be transmitted. As is well-known in the art, sync symbols allow the receiver to determine exactly when data symbols and pilot symbols will arrive, so that the input signal (or input symbol stream, as it is sometimes termed) may be sampled in exactly the proper places. Pilot symbols, which are predetermined symbols, are used to measure and compensate for the effects of the channel on the data symbols themselves.
Although this system tends to work well in the presence of frequency-selective fading, there are some problems that merit consideration. Even though the sub-carriers used to form the sub-channels may quite easily be started with a known phase at the beginning of the first time slot, by the beginning of the subsequent time slot the phase will generally be different. This means that, even if the same sync symbols and pilot symbols are used for each time slot, the sync and pilot symbols received will be different for every time slot by virtue of this sub-carrier phase shift.
An important consequence of this phase shift is that sync signal detection cannot be accomplished prior to demodulation without adding significant complexity. If the sync signals can be guaranteed to be identical at the composite signal level for every time slot, then a single matched filter can be used for sync timing detection.
A similar problem affects the ability to use pilot symbols to characterize channel effects. An optimum system should have the ability to detect a variety of pilot symbols, although a very large set of pilot symbols introduces an undesirable decoding burden. Since peak-to-average power considerations are important in the design of power amplifiers (PA's) for linear systems, it is desirable to retain the ability to use elements of a fairly large set of pilot symbols. This is because the pilot symbols represent a deterministic portion of the symbol stream, and pilot symbol values can be specifically selected to minimize peak-to-average power.
Accordingly, a need arises for a method for compensating for sub-carrier phase shift without adding unnecessary complexity to the system under consideration.